Time-Varying Gain Audio Processing for Fluctuating Noise
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Solution Overview
Problem
Hearing impaired listeners require a higher speech-to-noise ratio to understand speech in fluctuating noise environments, and conventional audio processing methods, including compression amplification, do not significantly improve intelligibility beyond what is achievable with linear-gain amplification.
Innovation Solution
The implementation of 'energy equalization' in audio processing, which involves introducing a time-varying gain to amplify signal segments below the long-term average level, with a response time of 5 ms and a long-term average duration of 200 ms, to match the signal level to the long-term average, thereby improving perception of weak sounds and reducing dynamic range.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Measurement precision
If compression amplification is used to amplify weak speech signals, then speech intelligibility for hearing impaired listeners is improved, but the dynamic range of the signal is reduced and performance in fluctuating noise does not exceed linear-gain amplification
Solution Approach 1:
The patent applies dynamic gain adjustment based on short-term signal level detection. The system continuously monitors the instantaneous energy of the speech signal and adapts the gain in real-time, switching between low-gain and high-gain states based on whether the signal exceeds a threshold. This dynamic adaptation allows the system to preserve transient speech components in fluctuating noise while maintaining overall intelligibility, resolving the contradiction between compression amplification benefits and performance in varying noise conditions.
Solution Approach 2:
The system performs preliminary detection of signal energy levels before applying gain amplification. By detecting whether the instantaneous signal energy exceeds a predetermined threshold before amplification, the system can preemptively apply appropriate gain levels. This preliminary action ensures that weak speech signals are amplified before they are lost in noise, while preventing over-amplification of already sufficient signals, thus improving speech intelligibility without sacrificing adaptability to fluctuating noise.
2Device complexity
If fast-attack and slow-release compression is applied over multiple syllables, then dynamic range compression is achieved, but the response time is too slow to capture transient speech components
Solution Approach 1:
The patent segments the speech signal processing into distinct temporal phases: detection phase, decision phase, and amplification phase. By dividing the processing into these segments with different time constants, the system achieves both fast response for transients and smooth overall compression. The detection phase operates with fast time constant to capture transients, while the amplification phase uses slower adjustment to maintain stability, thus resolving the contradiction between processing complexity and response speed.
Solution Approach 2:
The system changes the gain parameter dynamically based on the detected signal level. Instead of using fixed compression parameters, the gain is adjusted as a variable parameter that switches between predetermined values based on real-time signal detection. This parameter change approach allows the system to achieve fast response times by switching gain states instantly when transients are detected, while maintaining the overall compression effect through cumulative gain adjustments, thus resolving the speed-complexity contradiction.
Data Source
AI summary
An approach to audio processing aims to improve intelligibility by amplifying time segments of an input signal when the level of the signal falls below a long-term average level of the input signal, for instance, introducing a time-varying gain such that the signal level of the amplified segment matches the long-term average level.